Move ownership of receive ViEChannel to VideoReceiveStream.

webrtc/video/video_receive_stream.cc

 /*
  *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
 #include "webrtc/video/video_receive_stream.h"
 
 #include <stdlib.h>
 
 #include <string>
 
 #include "webrtc/base/checks.h"
 #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
 #include "webrtc/system_wrappers/interface/clock.h"
 #include "webrtc/system_wrappers/interface/logging.h"
 #include "webrtc/video/receive_statistics_proxy.h"
 #include "webrtc/video_encoder.h"
+#include "webrtc/video_engine/call_stats.h"
 #include "webrtc/video_receive_stream.h"
 
 namespace webrtc {
+
+static bool UseSendSideBwe(const std::vector<RtpExtension>& extensions) {
+  for (const auto& extension : extensions) {
+    if (extension.name == RtpExtension::kTransportSequenceNumber)
+      return true;
+  }
+  return false;
+}
+
 std::string VideoReceiveStream::Decoder::ToString() const {
   std::stringstream ss;
   ss << "{decoder: " << (decoder != nullptr ? "(VideoDecoder)" : "nullptr");
   ss << ", payload_type: " << payload_type;
   ss << ", payload_name: " << payload_name;
   ss << ", is_renderer: " << (is_renderer ? "yes" : "no");
   ss << ", expected_delay_ms: " << expected_delay_ms;
   ss << '}';
 
   return ss.str();
@@ skipping 89 matching lines @@
       Call::Config::kDefaultStartBitrateBps / 1000;
 
   return codec;
 }
 }  // namespace
 
 VideoReceiveStream::VideoReceiveStream(int num_cpu_cores,
                                        ChannelGroup* channel_group,
                                        int channel_id,
                                        const VideoReceiveStream::Config& config,
-                                       webrtc::VoiceEngine* voice_engine)
+                                       webrtc::VoiceEngine* voice_engine,
+                                       ProcessThread* process_thread)
     : transport_adapter_(config.rtcp_send_transport),
       encoded_frame_proxy_(config.pre_decode_callback),
       config_(config),
       clock_(Clock::GetRealTimeClock()),
-      channel_group_(channel_group),
-      channel_id_(channel_id) {
+      channel_group_(channel_group) {
   LOG(LS_INFO) << "VideoReceiveStream: " << config_.ToString();
-  RTC_CHECK(channel_group_->CreateReceiveChannel(
-      channel_id_, &transport_adapter_, num_cpu_cores, config));
 
-  vie_channel_ = channel_group_->GetChannel(channel_id_);
+  bool send_side_bwe = UseSendSideBwe(config_.rtp.extensions);
+
+  RemoteBitrateEstimator* bitrate_estimator =
+      channel_group_->GetRemoteBitrateEstimator(send_side_bwe);
+
+  vie_channel_.reset(new ViEChannel(
+      num_cpu_cores, &transport_adapter_, process_thread,
+      channel_group_->GetRtcpIntraFrameObserver(),
+      channel_group_->GetBitrateController()->CreateRtcpBandwidthObserver(),
+      nullptr, bitrate_estimator,
+      channel_group_->GetCallStats()->rtcp_rtt_stats(), channel_group_->pacer(),
+      channel_group_->packet_router(), 1, false));
+
+  RTC_CHECK(vie_channel_->Init() == 0);
+
+  // Register the channel to receive stats updates.
+  channel_group_->GetCallStats()->RegisterStatsObserver(
+      vie_channel_->GetStatsObserver());
+
 
   // TODO(pbos): This is not fine grained enough...
   vie_channel_->SetProtectionMode(config_.rtp.nack.rtp_history_ms > 0, false,
                                   -1, -1);
   RTC_DCHECK(config_.rtp.rtcp_mode != RtcpMode::kOff)
       << "A stream should not be configured with RTCP disabled. This value is "
          "reserved for internal usage.";
   vie_channel_->SetRTCPMode(config_.rtp.rtcp_mode);
 
   RTC_DCHECK(config_.rtp.remote_ssrc != 0);
@@ skipping 11 matching lines @@
     vie_channel_->SetRemoteSSRCType(kViEStreamTypeRtx, it->second.ssrc);
     vie_channel_->SetRtxReceivePayloadType(it->second.payload_type, it->first);
   }
   // TODO(holmer): When Chrome no longer depends on this being false by default,
   // always use the mapping and remove this whole codepath.
   vie_channel_->SetUseRtxPayloadMappingOnRestore(
       config_.rtp.use_rtx_payload_mapping_on_restore);
 
   // TODO(pbos): Remove channel_group_ usage from VideoReceiveStream. This
   // should be configured in call.cc.
-  channel_group_->SetChannelRembStatus(false, config_.rtp.remb, vie_channel_);
+  channel_group_->SetChannelRembStatus(false, config_.rtp.remb,
+                                       vie_channel_.get());
 
   for (size_t i = 0; i < config_.rtp.extensions.size(); ++i) {
     const std::string& extension = config_.rtp.extensions[i].name;
     int id = config_.rtp.extensions[i].id;
     // One-byte-extension local identifiers are in the range 1-14 inclusive.
     RTC_DCHECK_GE(id, 1);
     RTC_DCHECK_LE(id, 14);
     if (extension == RtpExtension::kTOffset) {
       RTC_CHECK_EQ(0, vie_channel_->SetReceiveTimestampOffsetStatus(true, id));
     } else if (extension == RtpExtension::kAbsSendTime) {
@@ skipping 70 matching lines @@
 
 VideoReceiveStream::~VideoReceiveStream() {
   LOG(LS_INFO) << "~VideoReceiveStream: " << config_.ToString();
   incoming_video_stream_->Stop();
   vie_channel_->RegisterPreRenderCallback(nullptr);
   vie_channel_->RegisterPreDecodeImageCallback(nullptr);
 
   for (size_t i = 0; i < config_.decoders.size(); ++i)
     vie_channel_->DeRegisterExternalDecoder(config_.decoders[i].payload_type);
 
-  channel_group_->DeleteChannel(channel_id_);
+  channel_group_->GetCallStats()->DeregisterStatsObserver(
+      vie_channel_->GetStatsObserver());
+  channel_group_->SetChannelRembStatus(false, false, vie_channel_.get());
+
+  uint32_t remote_ssrc = vie_channel_->GetRemoteSSRC();
+  bool send_side_bwe = UseSendSideBwe(config_.rtp.extensions);
+  channel_group_->GetRemoteBitrateEstimator(send_side_bwe)->RemoveStream(
+      remote_ssrc);
+  // This must be deleted before ReceiveStatisticsProxy, due to callbacks.

[pbos-webrtc, 2015/10/16 12:49:57]

Shouldn't ReceiveStatisticsProxy be declared before vie_channel_ then? I'd
rather enforce destruction order than add a reset().

[mflodman, 2015/10/16 13:31:57]

Done.

+  vie_channel_.reset();
 }
 
 void VideoReceiveStream::Start() {
   transport_adapter_.Enable();
   incoming_video_stream_->Start();
   vie_channel_->StartReceive();
 }
 
 void VideoReceiveStream::Stop() {
   incoming_video_stream_->Stop();
@@ skipping 52 matching lines @@
   return 0;
 }
 
 void VideoReceiveStream::SignalNetworkState(NetworkState state) {
   vie_channel_->SetRTCPMode(state == kNetworkUp ? config_.rtp.rtcp_mode
                                                 : RtcpMode::kOff);
 }
 
 }  // namespace internal
 }  // namespace webrtc